Posts Tagged ‘howto’
Using Dialplan Functions: AES_DECRYPT() and AES_ENCRYPT()
I recently asked on twitter how many people would be interested in a set of blog posts that focused on how to use the various dialplan functions in Asterisk, and I got quite a positive response. I posted that shortly before getting married, and now that I’m back into the groove of things, I’m going to take a shot at posting a bunch of content focused around Asterisk dialplan functions. If you don’t know what Asterisk dialplan functions are, head on over to the online version of Asterisk: The Definitive Guide (3rd edition) (or buy it) and read the section on dialplan functions. If you’re still starting out with Asterisk, I highly suggest you start with the dialplan basics chapter.
Today we’ll look at the first 2 dialplan functions in my list: AES_DECRYPT() and AES_ENCRYPT()
The AES_DECRYPT() and AES_ENCRYPT() functions work by passing strings to the functions, and they return a result. If you pass an unencrypted string to the AES_ENCRYPT() function it will return an encrypted string; vice-versa for the AES_DECRYPT() function. The two functions operate by passing a string and a key where the result is encoded in base64.
Use case for these functions probably makes the most sense when you need to store data outside of the dialplan, perhaps passwords, pins, or other data passed in by the caller, but which you want to secure when you go to store it. Let’s take an example where we create some dialplan that allows a caller to set their pin and store it in the database. For the sake of simplicity I’m not going to add any error checking (like to verify we really have data to work with, allow the caller to verify their extension, etc.):
exten => *88,1,NoOp()
same => n,Playback(silence/1)
same => n,Read(UserExtension,extension,3) ; read persons 3 digit extension unmber
same => n,Verbose(2,Extension number: ${UserExtension})
same => n,Read(PinEntry,agent-pass) ; ask for a pin number
same => n,Verbose(2,Pin number: ${PinEntry})
same => n,SayDigits(${PinEntry}) ; say pin back to caller
same => n,Set(DB(pin/${UserExtension})=${PinEntry}) ; store pin in the AstDB
same => n,Playback(vm-goodbye)
same => n,Hangup()
After the user enters their extension and pin, we store it in the AstDB. We can verify it was stored correctly by checking from the Asterisk CLI:
scrappy*CLI> database show pin
/pin/100 : 1234
1 results found.
Now let’s modify our dialplan to store the pin in the database using a value returned from AES_ENCRYPT().
exten => *88,1,NoOp()
same => n,Playback(silence/1)
same => n,Read(UserExtension,extension,3)
same => n,Verbose(2,Extension number: ${UserExtension})
same => n,Read(PinEntry,agent-pass)
same => n,Verbose(2,Pin number: ${PinEntry})
same => n,SayDigits(${PinEntry})
same => n,Set(SpecialKey=1234qwerasdfzxcv)
same => n,Set(EncryptedPin=${AES_ENCRYPT(${SpecialKey},${PinEntry})})
same => n,Set(DB(pin/${UserExtension})=${EncryptedPin})
same => n,Playback(vm-goodbye)
same => n,Hangup()
And we can see the encoded string stored in the database:
scrappy*CLI> database show pin
/pin/100 : Je2G/qyHuGVKgvvXDwXjHA==
1 results found.
Of course anyone who has access to the AstDB from the Asterisk CLI is also going to have access to the Asterisk dialplan, so you’ll have to do a better job than I have here of hiding the secret key being used for encrypting the data. Really all we’re trying to do here is not make the list of pins and data in our AstDB quite so obvious. We could of course not use AstDB at all, and store the data remotely where we know people will have access to the data, but not access to the secret key on our Asterisk server.
Now lets look at the inverse by decoding the pin to authenticate someone.
exten => *77,1,NoOp()
same => n,Playback(silence/1)
same => n,Read(UserExtension,extension,3) ; get users extension
same => n,Set(EncryptedPin=${DB(pin/${UserExtension})}) ; get encrypted pin from AstDB
same => n,Read(PinEntry,agent-pass) ; get pin from user
same => n,Set(SpecialKey=1234qwerasdfzxcv)
same => n,Set(DecryptedPin=${AES_DECRYPT(${SpecialKey},${EncryptedPin})}) ; decrypt the pin
same => n,Playback(${IF($["${PinEntry}" = "${DecryptedPin}"]?pin-number-accepted:pin-invalid)}) ; if pin is correct, play number accepted, else, pin invalid
same => n,Playback(vm-goodbye)
same => n,Hangup()
That’s it for now. Leave a comment if you like this format, and if you found this article useful. Thanks!
Scheduling automated calls between two participants with res_calendar
Here is a little dialplan snippet I wrote this morning for the next edition of the Asterisk book. While I’m not going to delve into all the aspects of setting up res_calendar like we do in the book, I thought for those of you who might already have this working might enjoy it.
(The calendaring modules are available in Asterisk 1.8, which is currently in release candidate status. Check out http://www.asterisk.org/downloads for the current version.)
I started with this little bit of dialplan that gets triggered when a call is answered from the calendaring module:
[AutomatedMeetingSetup]
exten => start,1,Verbose(2,Triggering meeting setup for two participants)
same => n,Set(DeviceToDial=${FILTER(0-9A-Za-z,${CALENDAR_EVENT(location)})})
same => n,Dial(SIP/${DeviceToDial},30)
same => n,Hangup()
The location field in my calendar event (which in this case is labeled as Destination in my Google Calendar) contains the string 0000FFFF0002 which is the device identifier in my example.
Once you get that working, the cool magic happens below. In the following example, a call is placed from the calendaring module when a meeting needs to take place between two participants. The first part of the dialplan allows the first person called to accept or reject the meeting, and if accepted, to record a message for the other party. Once that recording is saved, the dialplan will go ahead and trigger a call to the other meeting participant.
When that meeting participant answers the call, a Macro() is employed to allow them to listen to the recorded message left by the first party, (i.e. “Hey Jim, this is Leif. We have a meeting scheduled right now.”). That person is then presented the same option to accept or reject the call by pressing 1 or 2.
Of course the dialplan could even by further expanded to play back messages when the calls were rejected, the option for the called party to leave a return message of why they are rejecting the call, and maybe even the ability to post-pone the call for a few minutes. All it takes is some clever dialplan!
[AutomatedMeetingSetup]
exten => start,1,Verbose(2,Triggering meeting setup for two participants)
same => n,Read(CheckMeetingAcceptance,to-confirm-wakeup&press-1&otherwise&press-2,,1)
same => n,GotoIf($["${CheckMeetingAcceptance}" != "1"]?hangup,1)
same => n,Playback(silence/1&pls-rcrd-name-at-tone&and-prs-pound-whn-finished)
same => n,Set(__RandomNumber=${RAND()})
same => n,Record(/tmp/meeting-invite-${RandomNumber}.ulaw)
same => n,Set(DeviceToDial=${FILTER(0-9A-Za-z,${CALENDAR_EVENT(location)})})
same => n,Dial(SIP/${DeviceToDial},30,M(CheckConfirm))
same => n,Hangup()
exten => hangup,1,Verbose(2,Call was rejected)
same => n,Playback(vm-goodbye)
same => n,Hangup()
[macro-CheckConfirm]
exten => s,1,Verbose(2,Allowing called party to accept or reject)
same => n,Playback(/tmp/meeting-invite-${RandomNumber})
same => n,Read(CheckMeetingAcceptance,to-confirm-wakeup&press-1&otherwise&press-2,,1)
same => n,GotoIf($["${CheckMeetingAcceptance}" != "1"]?hangup,1)
exten => hangup,1,Verbose(2,Call was rejected by called party)
same => n,Playback(vm-goodbye)
same => n,Hangup()
Asterisk IMAP and Gmail
Today I was working on the next edition of the Asterisk book and wanted to see if I could get Asterisk IMAP voicemail support to work with Gmail. I had tried doing this a few times in the past without success, but since I had spent some time documenting and testing against Dovecot last week for another client and gotten everything working, I figured I had a good base to start trying to connect to the Gmail IMAP servers.
At first I was having problems with getting Asterisk to connect to the server as it would keep timing out when trying to connect to the Gmail IMAP servers on port 993. I looked at the instructions for connecting and double checked, and I thought I had everything right. I also made sure my IMAP library was compiled with OpenSSL support since Gmail requires you to connect via SSL.
After reading a few emails and doing more testing, I finally stumbled upon the missing link! Find below the snippet of voicemail.conf configuration that finally allowed me to connect to the Gmail IMAP system with Asterisk IMAP voicemail support.
Note: Be aware that there is a mixture of commas and pipes in the line where we’ve setup mailbox 100. This is not a typo!
; voicemail.conf imapserver=imap.gmail.com imapport=993 imapflags=ssl pollmailboxes=yes pollfreq=30 [default] 100 => 100,Leif Madsen,,,attach=no|imapuser=leif.madsen@MYDOMAIN_GOOGLE_APPS.com|imappassword=my_secret_password
I was using this with Google Apps for one of the domains we bought for working on the book, so the login is the full email address. The password is what you use for logging into the Gmail interface. Also, you need to make sure you’ve enabled IMAP support in the web interface before trying this or else your connection won’t work.
More information including step-by-step instructions on compiling IMAP support into Asterisk and configuration examples for Dovecot and Gmail will be in the upcoming book, but I was so excited to get this working today that I thought I’d share the secret sauce at least so you don’t have to wait for the whole recipe.
Connecting to Gmail was not as quick as connecting to Dovecot on a remote server I was working with. I imagine this is due to the high load Google has to deal with, so while it works, there may be some minor delays when retrieving and leaving voice messages, but nothing that caused it to be unusable.
Note: The KEY to making it work was the imapflags=ssl part. Without that you won’t get connected and will have problems with timeouts and such going on with Asterisk. Once I enabled the ssl flag I was golden.
Installing the Asterisk Test Suite
In case you missed it, I wrote an introductory article about the Asterisk Test Suite on the Asterisk blog. The post takes you from a minimally installed Ubuntu 9.10 up to the point that you can run the tests in the test suite by going through all the possible errors you might encounter getting it loaded, and how to resolve those errors (most of them being missing dependencies).
Enjoy!
http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/
Using a Nokia E71 with Asterisk (3G or WiFi)
There was some talk in one of the IRC chat rooms today about someone trying to get their E61i working with Asterisk. I haven’t had an issue getting that phone or my E71 phone working, but regardless he was having issues. I figured I might as well spend a little bit of time today going through my configuration, both for my own reference, and so that other people can get their Nokia’s setup with Asterisk as well.
I’ll be using the native SIP client, although I’ve had just as good of luck using the Fring application. The advantage to the Fring application is that you can use it with Skype, along with multiple IM clients, and also video! I especially like that the application makes use of the video camera on the front of the phone so that you can use it as a videophone. Using the video on a phone like the iPhone or Nexus One seems useless to me (but I digress!).
First, lets get our Asterisk configuration setup in sip.conf. We’ll need to make sure we’ve setup a realm in sip.conf as our phone will require it. If you don’t, then the default realm is ‘asterisk‘.
sip.conf:
[general] realm=pbx.my_asterisk_box.com disallow=all allow=ulaw allow=alaw srvlookup=yes pedantic=yes maxexpiry=360 minexpiry=120 defaultexpirey=120 videosupport=yes [leifmadsen_cell] type=friend secret=super_secret_password context=devices nat=yes canreinvite=no qualify=no mailbox=100@default callerid=Leif Madsen <571> insecure=invite,port subscribecontext=subscriptions disallow=all allow=g729 allow=ulaw
Service profile: IETF
Default access point: Select either a wifi connection or 3G connection. In my case I’m selecting “Rogers Internet”
Public user name: sip:leifmadsen_cell@pbx.my_asterisk_box.com (notice how leifmadsen_cell is the same as what we configured in sip.conf)
Use compression: No
Registration: Always on (you can set this to ‘When needed’ if you only want to place outbound calls via VoIP sometimes)
Use security: No
Proxy server address: pbx.my_asterisk_box.com
Realm: pbx.my_asterisk_box.com
Username: leifmadsen_cell
Password: super_secret_password
Allow loose routing: Yes
Transport type: UDP
Port: 5060 (unless you’ve changed this yourself on your Asterisk box)
(fill this information out exactly like the Proxy server section)
INVITE sip:8500@pbx.my_asterisk_server.com;user=phone SIP/2.0 Route: Via: SIP/2.0/UDP 10.10.10.84:5060;branch=z9hG4bKlq60dckmalhc6vap06nosen;rport From: ;tag=mh5gdciapphc6m6506no To: Contact: Supported: 100rel,sec-agree CSeq: 1252 INVITE Call-ID: rdw6Iy8zoIfKxg6LzJ7FSPdgBvIb8y Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Expires: 120 Privacy: none User-Agent: E71-2 RM-346 400.21.013 P-Preferred-Identity: sip:leifmadsen_cell@pbx.my_asterisk_server.com Max-Forwards: 70 Content-Type: application/sdp Accept: application/sdp Content-Length: 447 v=0 o=Nokia-SIPUA 63437257072703500 63437257072703500 IN IP4 10.10.10.84 s=- c=IN IP4 10.10.10.84 t=0 0 m=audio 49152 RTP/AVP 96 0 8 97 18 98 13 a=sendrecv a=ptime:20 a=maxptime:200 a=fmtp:96 mode-change-neighbor=1 a=fmtp:18 annexb=no a=fmtp:98 0-15 a=rtpmap:96 AMR/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:97 iLBC/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:98 telephone-event/8000/1 a=rtpmap:13 CN/8000/1
So beyond that, there shouldn’t be anything else you need to do. Using the same configuration in sip.conf for Asterisk should also work with Fring. Perhaps I’ll create another blog post in the future about using Fring with E71 if there is interest in that. Anyone who wants to try testing out some video calls through my Asterisk box using their Fring video enabled phone, just let me know offline and we’ll set something up!
Asterisk, and other worldly endeavours. 
You must be logged in to post a comment.